view libmpcodecs/ad_libdca.c @ 32860:77dd2bb3fd02

Make sure we don't store the same stats twice by nulling the first byte of the stats_out string once we have written it. I've verified that this is necessary, without it at normally at least the last stats line is duplicated, it could also happen when we call encode_video more often due to encoder delay as Reimar noted. Sample encode was with default -ovc lavc -lavcopts vpass=1: --- pass1stats.log 2011-02-21 15:44:42.314259000 +0100 +++ pass1stats.log.dedup_patch 2011-02-21 15:41:51.262778000 +0100 @@ -6421,4 +6421,3 @@ in:6420 out:6420 type:2 q:239 itex:0 ptex:15905 mv:441 misc:1911 fcode:1 bcode:1 mc-var:989 var:350603 icount:0 skipcount:373 hbits:55; in:6421 out:6421 type:2 q:247 itex:0 ptex:13020 mv:422 misc:1863 fcode:1 bcode:1 mc-var:953 var:352607 icount:0 skipcount:379 hbits:55; in:6422 out:6422 type:2 q:252 itex:0 ptex:3162 mv:258 misc:1293 fcode:1 bcode:1 mc-var:837 var:352872 icount:0 skipcount:449 hbits:55; -in:6422 out:6422 type:2 q:252 itex:0 ptex:3162 mv:258 misc:1293 fcode:1 bcode:1 mc-var:837 var:352872 icount:0 skipcount:449 hbits:55;

/*
 * DTS Coherent Acoustics stream decoder using libdca
 * This file is partially based on dtsdec.c r9036 from FFmpeg and ad_liba52.c
 *
 * Copyright (C) 2007 Roberto Togni
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <assert.h>
#include "config.h"

#include "mp_msg.h"
#include "ad_internal.h"
#include "dec_audio.h"

#include <dts.h>

static const ad_info_t info =
{
    "DTS decoding with libdca",
    "libdca",
    "Roberto Togni",
    "",
    ""
};

LIBAD_EXTERN(libdca)

#define DTSBUFFER_SIZE 18726
#define HEADER_SIZE 14

#define CONVERT_LEVEL 1
#define CONVERT_BIAS 0

static const char ch2flags[6] = {
    DTS_MONO,
    DTS_STEREO,
    DTS_3F,
    DTS_2F2R,
    DTS_3F2R,
    DTS_3F2R | DTS_LFE
};

static inline int16_t convert(sample_t s)
{
    int i = s * 0x7fff;

    return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}

static void convert2s16_multi(sample_t *f, int16_t *s16, int flags, int ch_out)
{
    int i;

    switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
    case DTS_MONO:
        if (ch_out == 1)
            for(i = 0; i < 256; i++)
                s16[i] = convert(f[i]);
        else
            for(i = 0; i < 256; i++){
                s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
                s16[5*i+4] = convert(f[i]);
            }
        break;
    case DTS_CHANNEL:
    case DTS_STEREO:
    case DTS_DOLBY:
        for(i = 0; i < 256; i++){
            s16[2*i] = convert(f[i]);
            s16[2*i+1] = convert(f[i+256]);
        }
        break;
    case DTS_3F:
        for(i = 0; i < 256; i++){
            s16[3*i] = convert(f[i+256]);
            s16[3*i+1] = convert(f[i+512]);
            s16[3*i+2] = convert(f[i]);
        }
        break;
    case DTS_2F2R:
        for(i = 0; i < 256; i++){
            s16[4*i] = convert(f[i]);
            s16[4*i+1] = convert(f[i+256]);
            s16[4*i+2] = convert(f[i+512]);
            s16[4*i+3] = convert(f[i+768]);
        }
        break;
    case DTS_3F2R:
        for(i = 0; i < 256; i++){
            s16[5*i] = convert(f[i+256]);
            s16[5*i+1] = convert(f[i+512]);
            s16[5*i+2] = convert(f[i+768]);
            s16[5*i+3] = convert(f[i+1024]);
            s16[5*i+4] = convert(f[i]);
        }
        break;
    case DTS_MONO | DTS_LFE:
        for(i = 0; i < 256; i++){
            s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
            s16[6*i+4] = convert(f[i]);
            s16[6*i+5] = convert(f[i+256]);
        }
        break;
    case DTS_CHANNEL | DTS_LFE:
    case DTS_STEREO | DTS_LFE:
    case DTS_DOLBY | DTS_LFE:
        for(i = 0; i < 256; i++){
            s16[6*i] = convert(f[i]);
            s16[6*i+1] = convert(f[i+256]);
            s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
            s16[6*i+5] = convert(f[i+512]);
        }
        break;
    case DTS_3F | DTS_LFE:
        for(i = 0; i < 256; i++){
            s16[6*i] = convert(f[i+256]);
            s16[6*i+1] = convert(f[i+512]);
            s16[6*i+2] = s16[6*i+3] = 0;
            s16[6*i+4] = convert(f[i]);
            s16[6*i+5] = convert(f[i+768]);
        }
        break;
    case DTS_2F2R | DTS_LFE:
        for(i = 0; i < 256; i++){
            s16[6*i] = convert(f[i]);
            s16[6*i+1] = convert(f[i+256]);
            s16[6*i+2] = convert(f[i+512]);
            s16[6*i+3] = convert(f[i+768]);
            s16[6*i+4] = 0;
            s16[6*i+5] = convert(f[1024]);
        }
        break;
    case DTS_3F2R | DTS_LFE:
        for(i = 0; i < 256; i++){
            s16[6*i] = convert(f[i+256]);
            s16[6*i+1] = convert(f[i+512]);
            s16[6*i+2] = convert(f[i+768]);
            s16[6*i+3] = convert(f[i+1024]);
            s16[6*i+4] = convert(f[i]);
            s16[6*i+5] = convert(f[i+1280]);
        }
        break;
    }
}

static void channels_info(int flags)
{
    int lfe = 0;
    char lfestr[5] = "";

    if (flags & DTS_LFE) {
        lfe = 1;
        strcpy(lfestr, "+lfe");
    }
    mp_msg(MSGT_DECAUDIO, MSGL_V, "DTS: ");
    switch(flags & DTS_CHANNEL_MASK){
    case DTS_MONO:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "1.%d (mono%s)", lfe, lfestr);
        break;
    case DTS_CHANNEL:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "2.%d (channel%s)", lfe, lfestr);
        break;
    case DTS_STEREO:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "2.%d (stereo%s)", lfe, lfestr);
        break;
    case DTS_3F:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "3.%d (3f%s)", lfe, lfestr);
        break;
    case DTS_2F2R:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "4.%d (2f+2r%s)", lfe, lfestr);
        break;
    case DTS_3F2R:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "5.%d (3f+2r%s)", lfe, lfestr);
        break;
    default:
        mp_msg(MSGT_DECAUDIO, MSGL_V, "x.%d (unknown%s)", lfe, lfestr);
    }
    mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
}

static int dts_sync(sh_audio_t *sh, int *flags)
{
    dts_state_t *s = sh->context;
    int length;
    int sample_rate;
    int frame_length;
    int bit_rate;

    sh->a_in_buffer_len=0;

    while(1) {
        while(sh->a_in_buffer_len < HEADER_SIZE) {
            int c = demux_getc(sh->ds);

            if(c < 0)
                return -1;
            sh->a_in_buffer[sh->a_in_buffer_len++] = c;
        }

        length = dts_syncinfo(s, sh->a_in_buffer, flags, &sample_rate,
                              &bit_rate, &frame_length);

        if(length >= HEADER_SIZE)
            break;

//        mp_msg(MSGT_DECAUDIO, MSGL_V, "skip\n");
        memmove(sh->a_in_buffer, sh->a_in_buffer+1, HEADER_SIZE-1);
        --sh->a_in_buffer_len;
    }

    demux_read_data(sh->ds, sh->a_in_buffer + HEADER_SIZE, length - HEADER_SIZE);

    sh->samplerate = sample_rate;
    sh->i_bps = bit_rate/8;

    return length;
}

static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen, int maxlen)
{
    dts_state_t *s = sh->context;
    int16_t *out_samples = (int16_t*)buf;
    int flags;
    level_t level;
    sample_t bias;
    int nblocks;
    int i;
    int data_size = 0;

    if(!sh->a_in_buffer_len)
        if(dts_sync(sh, &flags) < 0) return -1; /* EOF */
    sh->a_in_buffer_len=0;

    flags &= ~(DTS_CHANNEL_MASK | DTS_LFE);
    flags |= ch2flags[sh->channels - 1];

    level = CONVERT_LEVEL;
    bias = CONVERT_BIAS;
    flags |= DTS_ADJUST_LEVEL;
    if(dts_frame(s, sh->a_in_buffer, &flags, &level, bias)) {
        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_frame() failed\n");
        goto end;
    }

    nblocks = dts_blocks_num(s);

    for(i = 0; i < nblocks; i++) {
        if(dts_block(s)) {
            mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_block() failed\n");
            goto end;
        }

        convert2s16_multi(dts_samples(s), out_samples, flags, sh->channels);

        out_samples += 256 * sh->channels;
        data_size += 256 * sizeof(int16_t) * sh->channels;
    }

end:
    return data_size;
}

static int preinit(sh_audio_t *sh)
{
    /* 256 = samples per block, 16 = max number of blocks */
    sh->audio_out_minsize = audio_output_channels * sizeof(int16_t) * 256 * 16;
    sh->audio_in_minsize = DTSBUFFER_SIZE;
    sh->samplesize=2;

    return 1;
}

static int init(sh_audio_t *sh)
{
    dts_state_t *s;
    int flags;
    int decoded_bytes;

    s = dts_init(0);
    if(s == NULL) {
        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts_init() failed\n");
        return 0;
    }
    sh->context = s;

    if(dts_sync(sh, &flags) < 0) {
        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts sync failed\n");
        dts_free(s);
        return 0;
    }
    channels_info(flags);

    assert(audio_output_channels >= 1 && audio_output_channels <= 6);
    sh->channels = audio_output_channels;

    decoded_bytes = decode_audio(sh, sh->a_buffer, 1, sh->a_buffer_size);
    if(decoded_bytes > 0)
        sh->a_buffer_len = decoded_bytes;
    else {
        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "dts decode failed on first frame (up/downmix problem?)\n");
        dts_free(s);
        return 0;
    }

    return 1;
}

static void uninit(sh_audio_t *sh)
{
    dts_state_t *s = sh->context;

    dts_free(s);
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    int flags;

    switch(cmd){
        case ADCTRL_RESYNC_STREAM:
            dts_sync(sh, &flags);
            return CONTROL_TRUE;
    }
    return CONTROL_UNKNOWN;
}